Move initialization of WebRtcSessionDescriptionFactory to SdpHandler
Also move ssrc_generator and audio/video options, as well as some
signal handling that's related.
These variables were not referenced in peer_connection.cc any more.
Bug: webrtc:11995
Change-Id: I29f8661afad488380d256220b35330233e8233e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189967
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32471}
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 8a72bcf..e6e8c7f 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -13,23 +13,22 @@
#include <limits.h>
#include <stddef.h>
#include <algorithm>
-#include <limits>
#include <memory>
-#include <queue>
#include <set>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "api/jsep_ice_candidate.h"
+#include "api/rtp_parameters.h"
#include "api/rtp_transceiver_direction.h"
#include "api/task_queue/queued_task.h"
#include "api/transport/webrtc_key_value_config.h"
#include "api/uma_metrics.h"
-#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video/video_codec_constants.h"
#include "call/audio_state.h"
#include "call/packet_receiver.h"
+#include "media/base/media_channel.h"
#include "media/base/media_config.h"
#include "media/base/rid_description.h"
#include "media/base/stream_params.h"
@@ -40,11 +39,11 @@
#include "p2p/base/p2p_constants.h"
#include "p2p/base/p2p_transport_channel.h"
#include "p2p/base/transport_info.h"
-#include "pc/audio_rtp_receiver.h"
#include "pc/ice_server_parsing.h"
+#include "pc/rtp_receiver.h"
+#include "pc/rtp_sender.h"
#include "pc/sctp_transport.h"
#include "pc/simulcast_description.h"
-#include "pc/video_rtp_receiver.h"
#include "pc/webrtc_session_description_factory.h"
#include "rtc_base/bind.h"
#include "rtc_base/helpers.h"
@@ -54,11 +53,11 @@
#include "rtc_base/net_helper.h"
#include "rtc_base/network_constants.h"
#include "rtc_base/robo_caller.h"
-#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/task_utils/to_queued_task.h"
#include "rtc_base/trace_event.h"
+#include "rtc_base/unique_id_generator.h"
#include "system_wrappers/include/metrics.h"
using cricket::ContentInfo;
@@ -417,7 +416,6 @@
observer_ = dependencies.observer;
async_resolver_factory_ = std::move(dependencies.async_resolver_factory);
port_allocator_ = std::move(dependencies.allocator);
- packet_socket_factory_ = std::move(dependencies.packet_socket_factory);
ice_transport_factory_ = std::move(dependencies.ice_transport_factory);
tls_cert_verifier_ = std::move(dependencies.tls_cert_verifier);
@@ -564,40 +562,6 @@
sdp_handler_.UpdateNegotiationNeeded();
});
- video_options_.screencast_min_bitrate_kbps =
- configuration.screencast_min_bitrate;
- audio_options_.combined_audio_video_bwe =
- configuration.combined_audio_video_bwe;
-
- audio_options_.audio_jitter_buffer_max_packets =
- configuration.audio_jitter_buffer_max_packets;
-
- audio_options_.audio_jitter_buffer_fast_accelerate =
- configuration.audio_jitter_buffer_fast_accelerate;
-
- audio_options_.audio_jitter_buffer_min_delay_ms =
- configuration.audio_jitter_buffer_min_delay_ms;
-
- audio_options_.audio_jitter_buffer_enable_rtx_handling =
- configuration.audio_jitter_buffer_enable_rtx_handling;
-
- auto webrtc_session_desc_factory =
- std::make_unique<WebRtcSessionDescriptionFactory>(
- signaling_thread(), channel_manager(), &sdp_handler_, session_id(),
- dtls_enabled_, std::move(dependencies.cert_generator), certificate,
- &ssrc_generator_);
- webrtc_session_desc_factory->SignalCertificateReady.connect(
- this, &PeerConnection::OnCertificateReady);
-
- if (options.disable_encryption) {
- webrtc_session_desc_factory->SetSdesPolicy(cricket::SEC_DISABLED);
- }
-
- webrtc_session_desc_factory->set_enable_encrypted_rtp_header_extensions(
- GetCryptoOptions().srtp.enable_encrypted_rtp_header_extensions);
- webrtc_session_desc_factory->set_is_unified_plan(IsUnifiedPlan());
- sdp_handler_.SetSessionDescFactory(std::move(webrtc_session_desc_factory));
-
// Add default audio/video transceivers for Plan B SDP.
if (!IsUnifiedPlan()) {
rtp_manager()->transceivers()->Add(
@@ -610,6 +574,8 @@
int delay_ms =
return_histogram_very_quickly_ ? 0 : REPORT_USAGE_PATTERN_DELAY_MS;
+ sdp_handler_.Initialize(configuration, &dependencies);
+
message_handler_.RequestUsagePatternReport(
[this]() {
RTC_DCHECK_RUN_ON(signaling_thread());
@@ -617,13 +583,6 @@
},
delay_ms);
- if (dependencies.video_bitrate_allocator_factory) {
- video_bitrate_allocator_factory_ =
- std::move(dependencies.video_bitrate_allocator_factory);
- } else {
- video_bitrate_allocator_factory_ =
- CreateBuiltinVideoBitrateAllocatorFactory();
- }
return true;
}
@@ -1734,42 +1693,6 @@
Observer()->OnIceSelectedCandidatePairChanged(event);
}
-void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
- MediaStreamInterface* stream) {
- if (IsClosed()) {
- return;
- }
- rtp_manager()->AddAudioTrack(track, stream);
- sdp_handler_.UpdateNegotiationNeeded();
-}
-
-void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
- MediaStreamInterface* stream) {
- if (IsClosed()) {
- return;
- }
- rtp_manager()->RemoveAudioTrack(track, stream);
- sdp_handler_.UpdateNegotiationNeeded();
-}
-
-void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
- MediaStreamInterface* stream) {
- if (IsClosed()) {
- return;
- }
- rtp_manager()->AddVideoTrack(track, stream);
- sdp_handler_.UpdateNegotiationNeeded();
-}
-
-void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
- MediaStreamInterface* stream) {
- if (IsClosed()) {
- return;
- }
- rtp_manager()->RemoveVideoTrack(track, stream);
- sdp_handler_.UpdateNegotiationNeeded();
-}
-
absl::optional<std::string> PeerConnection::GetDataMid() const {
RTC_DCHECK_RUN_ON(signaling_thread());
switch (data_channel_type()) {
@@ -2133,11 +2056,6 @@
return transport_controller_->NeedsIceRestart(content_name);
}
-void PeerConnection::OnCertificateReady(
- const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
- transport_controller_->SetLocalCertificate(certificate);
-}
-
void PeerConnection::OnTransportControllerConnectionState(
cricket::IceConnectionState state) {
switch (state) {
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index dd7a3b4a..9edc43a 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -378,12 +378,8 @@
}
cricket::PortAllocator* port_allocator() { return port_allocator_.get(); }
Call* call_ptr() { return call_ptr_; }
- rtc::UniqueRandomIdGenerator* ssrc_generator() { return &ssrc_generator_; }
- const cricket::AudioOptions& audio_options() { return audio_options_; }
- const cricket::VideoOptions& video_options() { return video_options_; }
- VideoBitrateAllocatorFactory* video_bitrate_allocator_factory() {
- return video_bitrate_allocator_factory_.get();
- }
+
+ ConnectionContext* context() { return context_.get(); }
cricket::DataChannelType data_channel_type() const;
void SetIceConnectionState(IceConnectionState new_state);
@@ -391,19 +387,6 @@
// Report the UMA metric SdpFormatReceived for the given remote offer.
void ReportSdpFormatReceived(const SessionDescriptionInterface& remote_offer);
- // Signals from MediaStreamObserver.
- void OnAudioTrackAdded(AudioTrackInterface* track,
- MediaStreamInterface* stream)
- RTC_RUN_ON(signaling_thread());
- void OnAudioTrackRemoved(AudioTrackInterface* track,
- MediaStreamInterface* stream)
- RTC_RUN_ON(signaling_thread());
- void OnVideoTrackAdded(VideoTrackInterface* track,
- MediaStreamInterface* stream)
- RTC_RUN_ON(signaling_thread());
- void OnVideoTrackRemoved(VideoTrackInterface* track,
- MediaStreamInterface* stream)
- RTC_RUN_ON(signaling_thread());
// Returns true if the PeerConnection is configured to use Unified Plan
// semantics for creating offers/answers and setting local/remote
@@ -554,11 +537,6 @@
const PeerConnectionInterface::RTCConfiguration& config) const;
- // Called when an RTCCertificate is generated or retrieved by
- // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
- void OnCertificateReady(
- const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
-
// Returns true and the TransportInfo of the given |content_name|
// from |description|. Returns false if it's not available.
static bool GetTransportDescription(
@@ -663,7 +641,6 @@
// is not injected. It should be required once chromium supplies it.
std::unique_ptr<AsyncResolverFactory> async_resolver_factory_
RTC_GUARDED_BY(signaling_thread());
- std::unique_ptr<rtc::PacketSocketFactory> packet_socket_factory_;
std::unique_ptr<cricket::PortAllocator>
port_allocator_; // TODO(bugs.webrtc.org/9987): Accessed on both
// signaling and network thread.
@@ -715,29 +692,10 @@
bool dtls_enabled_ RTC_GUARDED_BY(signaling_thread()) = false;
- // Member variables for caching global options.
- cricket::AudioOptions audio_options_ RTC_GUARDED_BY(signaling_thread());
- cricket::VideoOptions video_options_ RTC_GUARDED_BY(signaling_thread());
-
UsagePattern usage_pattern_ RTC_GUARDED_BY(signaling_thread());
bool return_histogram_very_quickly_ RTC_GUARDED_BY(signaling_thread()) =
false;
- // This object should be used to generate any SSRC that is not explicitly
- // specified by the user (or by the remote party).
- // The generator is not used directly, instead it is passed on to the
- // channel manager and the session description factory.
- rtc::UniqueRandomIdGenerator ssrc_generator_
- RTC_GUARDED_BY(signaling_thread());
-
- // A video bitrate allocator factory.
- // This can injected using the PeerConnectionDependencies,
- // or else the CreateBuiltinVideoBitrateAllocatorFactory() will be called.
- // Note that one can still choose to override this in a MediaEngine
- // if one wants too.
- std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
- video_bitrate_allocator_factory_;
-
DataChannelController data_channel_controller_;
// Machinery for handling messages posted to oneself
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 81df99a..d9337f4 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -27,6 +27,7 @@
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/uma_metrics.h"
+#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "media/base/codec.h"
#include "media/base/media_engine.h"
#include "media/base/rid_description.h"
@@ -36,6 +37,7 @@
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_description_factory.h"
#include "p2p/base/transport_info.h"
+#include "pc/connection_context.h"
#include "pc/data_channel_utils.h"
#include "pc/media_protocol_names.h"
#include "pc/media_stream.h"
@@ -49,11 +51,13 @@
#include "pc/simulcast_description.h"
#include "pc/stats_collector.h"
#include "pc/usage_pattern.h"
+#include "pc/webrtc_session_description_factory.h"
#include "rtc_base/bind.h"
#include "rtc_base/helpers.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/ref_counted_object.h"
+#include "rtc_base/rtc_certificate.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/string_encode.h"
@@ -944,6 +948,61 @@
SdpOfferAnswerHandler::~SdpOfferAnswerHandler() {}
+void SdpOfferAnswerHandler::Initialize(
+ const PeerConnectionInterface::RTCConfiguration& configuration,
+ PeerConnectionDependencies* dependencies) {
+ RTC_DCHECK_RUN_ON(signaling_thread());
+ video_options_.screencast_min_bitrate_kbps =
+ configuration.screencast_min_bitrate;
+ audio_options_.combined_audio_video_bwe =
+ configuration.combined_audio_video_bwe;
+
+ audio_options_.audio_jitter_buffer_max_packets =
+ configuration.audio_jitter_buffer_max_packets;
+
+ audio_options_.audio_jitter_buffer_fast_accelerate =
+ configuration.audio_jitter_buffer_fast_accelerate;
+
+ audio_options_.audio_jitter_buffer_min_delay_ms =
+ configuration.audio_jitter_buffer_min_delay_ms;
+
+ audio_options_.audio_jitter_buffer_enable_rtx_handling =
+ configuration.audio_jitter_buffer_enable_rtx_handling;
+
+ // Obtain a certificate from RTCConfiguration if any were provided (optional).
+ rtc::scoped_refptr<rtc::RTCCertificate> certificate;
+ if (!configuration.certificates.empty()) {
+ // TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of
+ // just picking the first one. The decision should be made based on the DTLS
+ // handshake. The DTLS negotiations need to know about all certificates.
+ certificate = configuration.certificates[0];
+ }
+
+ webrtc_session_desc_factory_ =
+ std::make_unique<WebRtcSessionDescriptionFactory>(
+ signaling_thread(), channel_manager(), this, pc_->session_id(),
+ pc_->dtls_enabled(), std::move(dependencies->cert_generator),
+ certificate, &ssrc_generator_);
+ webrtc_session_desc_factory_->SignalCertificateReady.connect(
+ this, &SdpOfferAnswerHandler::OnCertificateReady);
+
+ if (pc_->context()->options().disable_encryption) {
+ webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
+ }
+
+ webrtc_session_desc_factory_->set_enable_encrypted_rtp_header_extensions(
+ pc_->GetCryptoOptions().srtp.enable_encrypted_rtp_header_extensions);
+ webrtc_session_desc_factory_->set_is_unified_plan(IsUnifiedPlan());
+
+ if (dependencies->video_bitrate_allocator_factory) {
+ video_bitrate_allocator_factory_ =
+ std::move(dependencies->video_bitrate_allocator_factory);
+ } else {
+ video_bitrate_allocator_factory_ =
+ CreateBuiltinVideoBitrateAllocatorFactory();
+ }
+}
+
// ==================================================================
// Access to pc_ variables
cricket::ChannelManager* SdpOfferAnswerHandler::channel_manager() const {
@@ -986,6 +1045,11 @@
// ===================================================================
+void SdpOfferAnswerHandler::OnCertificateReady(
+ const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
+ transport_controller()->SetLocalCertificate(certificate);
+}
+
void SdpOfferAnswerHandler::PrepareForShutdown() {
RTC_DCHECK_RUN_ON(signaling_thread());
weak_ptr_factory_.InvalidateWeakPtrs();
@@ -2495,14 +2559,14 @@
local_streams_->AddStream(local_stream);
MediaStreamObserver* observer = new MediaStreamObserver(local_stream);
- observer->SignalAudioTrackAdded.connect(pc_,
- &PeerConnection::OnAudioTrackAdded);
+ observer->SignalAudioTrackAdded.connect(
+ this, &SdpOfferAnswerHandler::OnAudioTrackAdded);
observer->SignalAudioTrackRemoved.connect(
- pc_, &PeerConnection::OnAudioTrackRemoved);
- observer->SignalVideoTrackAdded.connect(pc_,
- &PeerConnection::OnVideoTrackAdded);
+ this, &SdpOfferAnswerHandler::OnAudioTrackRemoved);
+ observer->SignalVideoTrackAdded.connect(
+ this, &SdpOfferAnswerHandler::OnVideoTrackAdded);
observer->SignalVideoTrackRemoved.connect(
- pc_, &PeerConnection::OnVideoTrackRemoved);
+ this, &SdpOfferAnswerHandler::OnVideoTrackRemoved);
stream_observers_.push_back(std::unique_ptr<MediaStreamObserver>(observer));
for (const auto& track : local_stream->GetAudioTracks()) {
@@ -2546,6 +2610,42 @@
UpdateNegotiationNeeded();
}
+void SdpOfferAnswerHandler::OnAudioTrackAdded(AudioTrackInterface* track,
+ MediaStreamInterface* stream) {
+ if (pc_->IsClosed()) {
+ return;
+ }
+ rtp_manager()->AddAudioTrack(track, stream);
+ UpdateNegotiationNeeded();
+}
+
+void SdpOfferAnswerHandler::OnAudioTrackRemoved(AudioTrackInterface* track,
+ MediaStreamInterface* stream) {
+ if (pc_->IsClosed()) {
+ return;
+ }
+ rtp_manager()->RemoveAudioTrack(track, stream);
+ UpdateNegotiationNeeded();
+}
+
+void SdpOfferAnswerHandler::OnVideoTrackAdded(VideoTrackInterface* track,
+ MediaStreamInterface* stream) {
+ if (pc_->IsClosed()) {
+ return;
+ }
+ rtp_manager()->AddVideoTrack(track, stream);
+ UpdateNegotiationNeeded();
+}
+
+void SdpOfferAnswerHandler::OnVideoTrackRemoved(VideoTrackInterface* track,
+ MediaStreamInterface* stream) {
+ if (pc_->IsClosed()) {
+ return;
+ }
+ rtp_manager()->RemoveVideoTrack(track, stream);
+ UpdateNegotiationNeeded();
+}
+
RTCError SdpOfferAnswerHandler::Rollback(SdpType desc_type) {
auto state = signaling_state();
if (state != PeerConnectionInterface::kHaveLocalOffer &&
@@ -4511,7 +4611,7 @@
voice_channel = channel_manager()->CreateVoiceChannel(
pc_->call_ptr(), pc_->configuration()->media_config, rtp_transport,
signaling_thread(), mid, pc_->SrtpRequired(), pc_->GetCryptoOptions(),
- pc_->ssrc_generator(), pc_->audio_options());
+ &ssrc_generator_, audio_options());
}
if (!voice_channel) {
return nullptr;
@@ -4538,8 +4638,8 @@
video_channel = channel_manager()->CreateVideoChannel(
pc_->call_ptr(), pc_->configuration()->media_config, rtp_transport,
signaling_thread(), mid, pc_->SrtpRequired(), pc_->GetCryptoOptions(),
- pc_->ssrc_generator(), pc_->video_options(),
- pc_->video_bitrate_allocator_factory());
+ &ssrc_generator_, video_options(),
+ video_bitrate_allocator_factory_.get());
}
if (!video_channel) {
return nullptr;
@@ -4575,7 +4675,7 @@
channel_manager()->CreateRtpDataChannel(
pc_->configuration()->media_config, rtp_transport,
signaling_thread(), mid, pc_->SrtpRequired(),
- pc_->GetCryptoOptions(), pc_->ssrc_generator()));
+ pc_->GetCryptoOptions(), &ssrc_generator_));
}
if (!data_channel_controller()->rtp_data_channel()) {
return false;
diff --git a/pc/sdp_offer_answer.h b/pc/sdp_offer_answer.h
index 0880c1d..873c24b 100644
--- a/pc/sdp_offer_answer.h
+++ b/pc/sdp_offer_answer.h
@@ -22,6 +22,7 @@
#include <vector>
#include "absl/types/optional.h"
+#include "api/audio_options.h"
#include "api/candidate.h"
#include "api/jsep.h"
#include "api/jsep_ice_candidate.h"
@@ -36,6 +37,7 @@
#include "api/set_remote_description_observer_interface.h"
#include "api/transport/data_channel_transport_interface.h"
#include "api/turn_customizer.h"
+#include "media/base/media_channel.h"
#include "media/base/stream_params.h"
#include "p2p/base/port_allocator.h"
#include "pc/channel.h"
@@ -52,6 +54,7 @@
#include "pc/rtp_receiver.h"
#include "pc/rtp_sender.h"
#include "pc/rtp_transceiver.h"
+#include "pc/rtp_transmission_manager.h"
#include "pc/sctp_transport.h"
#include "pc/session_description.h"
#include "pc/stats_collector.h"
@@ -85,16 +88,17 @@
// - Parsing and interpreting SDP.
// - Generating offers and answers based on the current state.
// This class lives on the signaling thread.
-class SdpOfferAnswerHandler {
+class SdpOfferAnswerHandler : public sigslot::has_slots<> {
public:
explicit SdpOfferAnswerHandler(PeerConnection* pc);
~SdpOfferAnswerHandler();
- void SetSessionDescFactory(
- std::unique_ptr<WebRtcSessionDescriptionFactory> factory) {
- RTC_DCHECK_RUN_ON(signaling_thread());
- webrtc_session_desc_factory_ = std::move(factory);
- }
+ // Called from PeerConnection's Initialize() function. Can only be called
+ // once. Modifies dependencies.
+ void Initialize(
+ const PeerConnectionInterface::RTCConfiguration& configuration,
+ PeerConnectionDependencies* dependencies);
+
void ResetSessionDescFactory() {
RTC_DCHECK_RUN_ON(signaling_thread());
webrtc_session_desc_factory_.reset();
@@ -255,6 +259,20 @@
bool IsUnifiedPlan() const RTC_RUN_ON(signaling_thread());
+ // Signals from MediaStreamObserver.
+ void OnAudioTrackAdded(AudioTrackInterface* track,
+ MediaStreamInterface* stream)
+ RTC_RUN_ON(signaling_thread());
+ void OnAudioTrackRemoved(AudioTrackInterface* track,
+ MediaStreamInterface* stream)
+ RTC_RUN_ON(signaling_thread());
+ void OnVideoTrackAdded(VideoTrackInterface* track,
+ MediaStreamInterface* stream)
+ RTC_RUN_ON(signaling_thread());
+ void OnVideoTrackRemoved(VideoTrackInterface* track,
+ MediaStreamInterface* stream)
+ RTC_RUN_ON(signaling_thread());
+
// | desc_type | is the type of the description that caused the rollback.
RTCError Rollback(SdpType desc_type);
void OnOperationsChainEmpty();
@@ -526,6 +544,11 @@
// payload type based demuxing in the affected channels.
bool UpdatePayloadTypeDemuxingState(cricket::ContentSource source);
+ // Called when an RTCCertificate is generated or retrieved by
+ // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
+ void OnCertificateReady(
+ const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
+
// ==================================================================
// Access to pc_ variables
cricket::ChannelManager* channel_manager() const;
@@ -538,6 +561,8 @@
RtpTransmissionManager* rtp_manager();
const RtpTransmissionManager* rtp_manager() const;
// ===================================================================
+ const cricket::AudioOptions& audio_options() { return audio_options_; }
+ const cricket::VideoOptions& video_options() { return video_options_; }
PeerConnection* const pc_;
@@ -616,6 +641,25 @@
SessionError::kNone;
std::string session_error_desc_ RTC_GUARDED_BY(signaling_thread());
+ // Member variables for caching global options.
+ cricket::AudioOptions audio_options_ RTC_GUARDED_BY(signaling_thread());
+ cricket::VideoOptions video_options_ RTC_GUARDED_BY(signaling_thread());
+
+ // This object should be used to generate any SSRC that is not explicitly
+ // specified by the user (or by the remote party).
+ // The generator is not used directly, instead it is passed on to the
+ // channel manager and the session description factory.
+ rtc::UniqueRandomIdGenerator ssrc_generator_
+ RTC_GUARDED_BY(signaling_thread());
+
+ // A video bitrate allocator factory.
+ // This can be injected using the PeerConnectionDependencies,
+ // or else the CreateBuiltinVideoBitrateAllocatorFactory() will be called.
+ // Note that one can still choose to override this in a MediaEngine
+ // if one wants too.
+ std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
+ video_bitrate_allocator_factory_;
+
rtc::WeakPtrFactory<SdpOfferAnswerHandler> weak_ptr_factory_
RTC_GUARDED_BY(signaling_thread());
};
